Thank David and Neeraj for your input. Neeraj, I posted the configs in my first post, but i've also attached some extracts here. they haven't changed much.
David, You're absolutely right and i think the problem could be the reverse dial-peer or DTMF configuration. I think I have the corresponding reverse dial-peer and the DTMF conf that you said. However, I have checked my side and all seems to be ok. I've also tried changing the dtmfmode to sip-notify on the gateway (and info in sip.conf) but no luck! Please look at the attached and give me some pointers. Thanks, Tim On Sun, May 17, 2009 at 3:44 PM, David Backeberg <dbackeb...@gmail.com> wrote: > On Sat, May 16, 2009 at 10:22 AM, Timothy Smith <timotsm...@gmail.com> wrote: >> I have finally managed to get voice working. I both parties can hear >> each other. The problem was nating. Our network is fairly big and >> these machines are atleast 2 switches from each other. I just enabled >> it (nat=route or nat=yes) and it worked. >> >> It's not yet done however. When I redirect a call to any Asterisk >> application, it just hangs up! I have read some history and archives, >> but none of the solutions has worked for me. e.g ip inspect udp >> idle-time 900. My router (or IOS) doesn't have thet command. >> >> Could you please assist point to what could be causing this and how to >> solve it? Below are some logs and attached is the router log. >> >> ; This is the extension conf. Enter the extension you want to reach >> now (something like auto attendant). >> exten => _X.,1,Read(NUM,beep,4,2,3) >> exten => _X.,n,Dial(SIP/${NUM}) >> >> ; This is all i get when i call and the call hangs up! > > Did you ever set up that reverse dial-peer? If not, do that first. > > You put a three second timeout on the Read(). By any chance, is the > call hanging up 3 seconds after you call? That would be expected > behavior. Well, actually you give it two tries. So it should be > beep > three second wait > beep > three second wait > hangup > > If you're actually entering numbers on your dialpad and they're not > getting read, you have a misconfiguration on your DTMF. If you enable > sip debugging on your asterisk side you can see exactly what's coming > over the wire from the Cisco side. There are a lot of choices for DTMF > on the asterisk side and the Cisco side, and they need to agree for > the button presses to be encoded and passed correctly. You can pass > them in-line as real audio, or you can convert them to a special dtmf > sip encoding. You'll notice all those choices when you go to configure > the Cisco dial-peer. > > My personal preference: > on the Cisco dial-peer side > dtmf-relay rtp-nte > > on the asterisk side > I left the dtmf config blank, and I don't remember which default you > end up with, but it worked in the default config for me. > > _______________________________________________
interface Serial0/0/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn negotiate-bchan resend-setup no cdp enable ! interface Serial0/0/1:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice isdn negotiate-bchan resend-setup no cdp enable dial-peer voice 1 pots destination-pattern 0T port 0/0/1:15 forward-digits all ! dial-peer voice 3 pots incoming called-number . direct-inward-dial port 0/0/1:15 dial-peer voice 4 pots incoming called-number . direct-inward-dial port 0/0/1:15 ! dial-peer voice 112 voip destination-pattern 730732888 monitor probe icmp-ping session protocol sipv2 session target ipv4:172.19.3.150 session transport udp dtmf-relay rtp-nte codec g711ulaw VG2# show dial-peer voice 112 VoiceOverIpPeer112 peer type = voice, system default peer = FALSE, information type = voice, description = `', tag = 112, destination-pattern = `730732888', voice reg type = 0, corresponding tag = 0, allow watch = FALSE answer-address = `', preference=0, CLID Restriction = None CLID Network Number = `' CLID Second Number sent CLID Override RDNIS = disabled, source carrier-id = `', target carrier-id = `', source trunk-group-label = `', target trunk-group-label = `', numbering Type = `unknown' group = 112, Admin state is up, Operation state is up, incoming called-number = `', connections/maximum = 0/unlimited, DTMF Relay = enabled, modem transport = system, URI classes: Incoming (Request) = Incoming (To) = Incoming (From) = Destination = huntstop = disabled, in bound application associated: 'DEFAULT' out bound application associated: '' dnis-map = permission :both incoming COR list:maximum capability outgoing COR list:minimum requirement Translation profile (Incoming): Translation profile (Outgoing): incoming call blocking: translation-profile = `' disconnect-cause = `no-service' advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4 type = voip, session-target = `ipv4:172.19.3.150', technology prefix: settle-call = disabled ip media DSCP = ef, ip signaling DSCP = af31, ip video rsvp-none DSCP = af41,ip video rsvp-pass DSCP = af41 ip video rsvp-fail DSCP = af41, UDP checksum = disabled, session-protocol = sipv2, session-transport = udp, req-qos = best-effort, acc-qos = best-effort, req-qos video = best-effort, acc-qos video = best-effort, req-qos audio def bandwidth = 64, req-qos audio max bandwidth = 0, req-qos video def bandwidth = 384, req-qos video max bandwidth = 0, dtmf-relay = rtp-nte, RTP dynamic payload type values: NTE = 101 Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122 CAS=123, TTY=119, ClearChan=125, PCM switch over u-law=0, A-law=8, GSMAMR-NB=117 iLBC=116 h263+=118, h264=119 G726r16 using static payload G726r24 using static payload RTP comfort noise payload type = 19 fax rate = voice, payload size = 20 bytes fax protocol = system fax-relay ecm enable Fax Relay SG3-to-G3 Enabled (by system configuration) fax NSF = 0xAD0051 (default) codec = g711ulaw, payload size = 160 bytes, video codec = None voice class codec = `' text relay = disabled Media Setting = flow-through (global) Expect factor = 10, Icpif = 20, Playout Mode is set to adaptive, Initial 60 ms, Max 250 ms Playout-delay Minimum mode is set to default, value 40 ms Fax nominal 300 ms Max Redirects = 1, signaling-type = cas, VAD = enabled, Poor QOV Trap = disabled, Source Interface = NONE voice class sip url = system, voice class sip rel1xx = system, tvoice class sip outbound-proxy = system, voice class sip asserted-id = system, voice class sip privacy = system, voice class sip e911 = system, voice class sip authenticate redirecting-number = system, redirect ip2ip = disabled local peer = false monitor probe method: icmp-ping monitoring session target, Monitored destination reachable Secure RTP: system (use the global setting) voice class perm tag = `' Time elapsed since last clearing of voice call statistics never Connect Time = 32060, Charged Units = 0, Successful Calls = 86, Failed Calls = 3, Incomplete Calls = 0 Accepted Calls = 0, Refused Calls = 0, Last Disconnect Cause is "2F ", Last Disconnect Text is "no resource (47)", Last Setup Time = 378671190. Last Disconnect Time = 378671508.
[May 18 09:54:58] DEBUG[28907] channel.c: Avoiding initial deadlock for channel '0x893750' [May 18 09:54:58] DEBUG[28907] channel.c: Avoiding initial deadlock for channel '0x893750' [May 18 09:54:58] DEBUG[10017] pbx.c: Launching 'Set' [May 18 09:54:58] VERBOSE[10017] logger.c: -- Executing [730732...@default:1] Set("SIP/172.17.3.248-008c4790", "CALLERID(num)=730730199") in new stack [May 18 09:54:58] DEBUG[10017] pbx.c: Launching 'Wait' [May 18 09:54:58] VERBOSE[10017] logger.c: -- Executing [730732...@default:2] Wait("SIP/172.17.3.248-008c4790", "3") in new stack [May 18 09:55:01] DEBUG[10017] pbx.c: Launching 'Read' [May 18 09:55:01] VERBOSE[10017] logger.c: -- Executing [730732...@default:3] Read("SIP/172.17.3.248-008c4790", "NUM,beep,4,2,3") in new stack [May 18 09:55:01] VERBOSE[10017] logger.c: -- Accepting a maximum of 4 digits. [May 18 09:55:01] DEBUG[10017] chan_sip.c: SIP answering channel: SIP/172.17.3.248-008c4790 [May 18 09:55:01] DEBUG[10017] chan_sip.c: Setting framing from config on incoming call [May 18 09:55:01] DEBUG[10017] chan_sip.c: ** Our capability: 0x4 (ulaw) Video flag: True Text flag: True [May 18 09:55:01] DEBUG[10017] chan_sip.c: ** Our prefcodec: 0x0 (nothing) [May 18 09:55:01] VERBOSE[10017] logger.c: Audio is at 172.19.3.150 port 15200 [May 18 09:55:01] VERBOSE[10017] logger.c: Adding codec 0x4 (ulaw) to SDP [May 18 09:55:01] VERBOSE[10017] logger.c: Adding non-codec 0x1 (telephone-event) to SDP [May 18 09:55:01] DEBUG[28907] channel.c: Avoiding initial deadlock for channel '0x893750' [May 18 09:55:01] VERBOSE[10017] logger.c: <--- Reliably Transmitting (NAT) to 172.17.3.248:61939 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.3.248:5060;branch=z9hG4bK10518C0;received=172.17.3.248 From: <sip:730730...@172.17.3.248>;tag=E20FA500-53D To: <sip:730732...@172.19.3.150>;tag=as1a442e78 Call-ID: 7864a6e7-42af11de-b211c927-51af5...@172.17.3.248 CSeq: 101 INVITE User-Agent: Asterisk PBX 1.6.0.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:730732...@172.19.3.150> Content-Type: application/sdp Content-Length: 261 v=0 o=root 877709162 877709162 IN IP4 172.19.3.150 s=Asterisk PBX 1.6.0.9 c=IN IP4 172.19.3.150 t=0 0 m=audio 15200 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> [May 18 09:55:01] DEBUG[10017] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 172.17.3.248:61939 [May 18 09:55:01] VERBOSE[28920] logger.c: <--- SIP read from UDP://172.17.3.248:61939 ---> ACK sip:730732...@172.19.3.150:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.3.248:5060;branch=z9hG4bK106BE5 From: <sip:730730...@172.17.3.248>;tag=E20FA500-53D To: <sip:730732...@172.19.3.150>;tag=as1a442e78 Date: Mon, 18 May 2009 06:53:47 GMT Call-ID: 7864a6e7-42af11de-b211c927-51af5...@172.17.3.248 Max-Forwards: 70 CSeq: 101 ACK Allow-Events: telephone-event Content-Length: 0 <-------------> [May 18 09:55:01] VERBOSE[28920] logger.c: --- (10 headers 0 lines) --- [May 18 09:55:01] DEBUG[28920] chan_sip.c: Stopping retransmission on '7864a6e7-42af11de-b211c927-51af5...@172.17.3.248' of Response 101: Match Found [May 18 09:55:01] VERBOSE[28920] logger.c: <--- SIP read from UDP://172.17.3.248:61939 ---> BYE sip:730732...@172.19.3.150:5060 SIP/2.0 Via: SIP/2.0/UDP 172.17.3.248:5060;branch=z9hG4bK107F4B From: <sip:730730...@172.17.3.248>;tag=E20FA500-53D To: <sip:730732...@172.19.3.150>;tag=as1a442e78 Date: Mon, 18 May 2009 06:53:47 GMT Call-ID: 7864a6e7-42af11de-b211c927-51af5...@172.17.3.248 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 70 Timestamp: 1242629630 CSeq: 102 BYE Reason: Q.850;cause=47 Content-Length: 0 <-------------> [May 18 09:55:01] VERBOSE[28920] logger.c: --- (12 headers 0 lines) --- [May 18 09:55:01] DEBUG[28920] chan_sip.c: Initializing initreq for method BYE - callid 7864a6e7-42af11de-b211c927-51af5...@172.17.3.248 [May 18 09:55:01] VERBOSE[28920] logger.c: Sending to 172.17.3.248 : 61939 (NAT) [May 18 09:55:01] VERBOSE[28920] logger.c: <--- Transmitting (NAT) to 172.17.3.248:61939 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.17.3.248:5060;branch=z9hG4bK107F4B;received=172.17.3.248 From: <sip:730730...@172.17.3.248>;tag=E20FA500-53D To: <sip:730732...@172.19.3.150>;tag=as1a442e78 Call-ID: 7864a6e7-42af11de-b211c927-51af5...@172.17.3.248 CSeq: 102 BYE User-Agent: Asterisk PBX 1.6.0.9 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 <------------> [May 18 09:55:01] DEBUG[28920] chan_sip.c: Trying to put 'SIP/2.0 20' onto UDP socket destined for 172.17.3.248:61939 [May 18 09:55:01] DEBUG[10017] pbx.c: Extension 730732888, priority 3 returned normally even though call was hung up [May 18 09:55:01] DEBUG[10017] channel.c: Soft-Hanging up channel 'SIP/172.17.3.248-008c4790' [May 18 09:55:01] DEBUG[10017] channel.c: Hanging up channel 'SIP/172.17.3.248-008c4790' [May 18 09:55:01] DEBUG[10017] chan_sip.c: Hangup call SIP/172.17.3.248-008c4790, SIP callid 7864a6e7-42af11de-b211c927-51af5...@172.17.3.248 [May 18 09:55:01] VERBOSE[28920] logger.c: Really destroying SIP dialog '7864a6e7-42af11de-b211c927-51af5...@172.17.3.248' Method: BYE [May 18 09:55:14] VERBOSE[9223] logger.c: -- Remote UNIX connection disconnected [May 18 09:55:21] DEBUG[28920] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [May 18 09:55:21] DEBUG[28920] chan_sip.c: Initializing initreq for method OPTIONS - callid 6979bc010cad89f67f32e3b508075...@172.19.3.150 [May 18 09:55:21] VERBOSE[28920] logger.c: Reliably Transmitting (NAT) to 172.17.10.150:5060: OPTIONS sip:172.17.10.150 SIP/2.0 Via: SIP/2.0/UDP 172.19.3.150:5060;branch=z9hG4bK71cec630 Max-Forwards: 70 From: "asterisk" <sip:aster...@172.19.3.150>;tag=as55864c67 To: <sip:172.17.10.150> Contact: <sip:aster...@172.19.3.150> Call-ID: 6979bc010cad89f67f32e3b508075...@172.19.3.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.9 Date: Mon, 18 May 2009 06:55:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- [May 18 09:55:21] DEBUG[28920] chan_sip.c: Trying to put 'OPTIONS si' onto UDP socket destined for 172.17.10.150:5060 [May 18 09:55:21] DEBUG[28920] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [May 18 09:55:21] DEBUG[28920] chan_sip.c: Initializing initreq for method OPTIONS - callid 6a5999803ab2c8364e9d6c2a33726...@172.19.3.150 [May 18 09:55:21] VERBOSE[28920] logger.c: Reliably Transmitting (NAT) to 172.17.3.249:5060: OPTIONS sip:172.17.3.249 SIP/2.0 Via: SIP/2.0/UDP 172.19.3.150:5060;branch=z9hG4bK597b7843;rport Max-Forwards: 70 From: "asterisk" <sip:aster...@172.19.3.150>;tag=as09b5f074 To: <sip:172.17.3.249> Contact: <sip:aster...@172.19.3.150> Call-ID: 6a5999803ab2c8364e9d6c2a33726...@172.19.3.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.9 Date: Mon, 18 May 2009 06:55:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- [May 18 09:55:21] DEBUG[28920] chan_sip.c: Trying to put 'OPTIONS si' onto UDP socket destined for 172.17.3.249:5060 [May 18 09:55:21] DEBUG[28920] chan_sip.c: Allocating new SIP dialog for (No Call-ID) - OPTIONS (No RTP) [May 18 09:55:21] DEBUG[28920] chan_sip.c: Initializing initreq for method OPTIONS - callid 512cece90ef7874550fd7bc45676c...@172.19.3.150 [May 18 09:55:21] VERBOSE[28920] logger.c: Reliably Transmitting (NAT) to 172.17.3.248:5060: OPTIONS sip:172.17.3.248 SIP/2.0 Via: SIP/2.0/UDP 172.19.3.150:5060;branch=z9hG4bK2b7ca253 Max-Forwards: 70 From: "asterisk" <sip:aster...@172.19.3.150>;tag=as2cf79799 To: <sip:172.17.3.248> Contact: <sip:aster...@172.19.3.150> Call-ID: 512cece90ef7874550fd7bc45676c...@172.19.3.150 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.0.9 Date: Mon, 18 May 2009 06:55:21 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Length: 0 --- [May 18 09:55:21] DEBUG[28920] chan_sip.c: Trying to put 'OPTIONS si' onto UDP socket destined for 172.17.3.248:5060 [May 18 09:55:21] VERBOSE[28920] logger.c: <--- SIP read from UDP://172.17.10.150:5060 ---> SIP/2.0 400 Bad Request - 'Malformed/Missing URL' Via: SIP/2.0/UDP 172.19.3.150:5060;branch=z9hG4bK71cec630 From: "asterisk" <sip:aster...@172.19.3.150>;tag=as55864c67 To: <sip:172.17.10.150> Call-ID: 6979bc010cad89f67f32e3b508075...@172.19.3.150 CSeq: 102 OPTIONS Content-Length: 0 <-------------> [May 18 09:55:21] VERBOSE[28920] logger.c: --- (7 headers 0 lines) --- [May 18 09:55:21] DEBUG[28920] chan_sip.c: Stopping retransmission on '6979bc010cad89f67f32e3b508075...@172.19.3.150' of Request 102: Match Found [May 18 09:55:21] VERBOSE[28920] logger.c: Really destroying SIP dialog '6979bc010cad89f67f32e3b508075...@172.19.3.150' Method: OPTIONS [May 18 09:55:21] VERBOSE[28920] logger.c: <--- SIP read from UDP://172.17.3.249:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.19.3.150:5060;branch=z9hG4bK597b7843;rport From: "asterisk" <sip:aster...@172.19.3.150>;tag=as09b5f074 To: <sip:172.17.3.249>;tag=6114EB94-1696 Date: Mon, 18 May 2009 06:53:53 GMT Call-ID: 6a5999803ab2c8364e9d6c2a33726...@172.19.3.150 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 OPTIONS Supported: 100rel,resource-priority,replaces Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Accept: application/sdp Content-Type: application/sdp Content-Length: 166 v=0 o=CiscoSystemsSIP-GW-UserAgent 3560 8246 IN IP4 172.19.2.41 s=SIP Call c=IN IP4 172.17.3.249 t=0 0 m=audio 0 RTP/AVP 18 0 8 4 2 15 3 c=IN IP4 172.17.3.249 <-------------> [May 18 09:55:21] VERBOSE[28920] logger.c: --- (14 headers 7 lines) --- [May 18 09:55:21] DEBUG[28920] chan_sip.c: Stopping retransmission on '6a5999803ab2c8364e9d6c2a33726...@172.19.3.150' of Request 102: Match Found [May 18 09:55:21] VERBOSE[28920] logger.c: Really destroying SIP dialog '6a5999803ab2c8364e9d6c2a33726...@172.19.3.150' Method: OPTIONS [May 18 09:55:21] VERBOSE[28920] logger.c: <--- SIP read from UDP://172.17.3.248:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 172.19.3.150:5060;branch=z9hG4bK2b7ca253 From: "asterisk" <sip:aster...@172.19.3.150>;tag=as2cf79799 To: <sip:172.17.3.248>;tag=E2100140-13C1 Date: Mon, 18 May 2009 06:54:11 GMT Call-ID: 512cece90ef7874550fd7bc45676c...@172.19.3.150 Server: Cisco-SIPGateway/IOS-12.x CSeq: 102 OPTIONS Supported: 100rel,resource-priority,replaces Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Accept: application/sdp Content-Type: application/sdp Content-Length: 166 v=0 o=CiscoSystemsSIP-GW-UserAgent 1387 515 IN IP4 172.17.3.248 s=SIP Call c=IN IP4 172.17.3.248 t=0 0 m=audio 0 RTP/AVP 18 0 8 4 2 15 3 c=IN IP4 172.17.3.248 <-------------> [May 18 09:55:21] VERBOSE[28920] logger.c: --- (14 headers 7 lines) --- [May 18 09:55:21] DEBUG[28920] chan_sip.c: Stopping retransmission on '512cece90ef7874550fd7bc45676c...@172.19.3.150' of Request 102: Match Found [May 18 09:55:21] VERBOSE[28920] logger.c: Really destroying SIP dialog '512cece90ef7874550fd7bc45676c...@172.19.3.150' Method: OPTIONS
May 18 06:57:22.762: ISDN Se0/0/1:15 Q931: RX <- SETUP pd = 8 callref = 0x0E82 Sending Complete Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98385 Exclusive, Channel 5 Calling Party Number i = 0x2183, '730730199' Plan:ISDN, Type:National Called Party Number i = 0xA1, '730732888' Plan:ISDN, Type:National May 18 06:57:22.770: ISDN Se0/0/1:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8E82 Channel ID i = 0xA98385 Exclusive, Channel 5 OULKLAVG2# May 18 06:57:25.778: ISDN Se0/0/1:15 Q931: TX -> CONNECT pd = 8 callref = 0x8E82 May 18 06:57:25.790: ISDN Se0/0/1:15 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x0E82 May 18 06:57:25.790: %ISDN-6-CONNECT: Interface Serial0/0/1:4 is now connected to 730730199 N/A May 18 06:57:25.930: ISDN Se0/0/1:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x0E82 Cause i = 0x82AF - Resource unavailable, unspecified May 18 06:57:25.930: %ISDN-6-CONNECT: Interface Serial0/0/1:4 is now connected to 730730199 N/A May 18 06:57:25.930: ISDN Se0/0/1:15 Q931: TX -> RELEASE pd = 8 callref = 0x8E82 May 18 06:57:25.942: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0E82 Cause i = 0x80D1 - Invalid call reference value May 18 06:57:25.954: ISDN Se0/0/1:15 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x0E82 May 18 06:57:25.954: ISDN Se0/0/1:15 **ERROR**: Ux_BadMsg: Invalid Message for call state 19, call id 0x3611, call ref 0x8E82, event 0xF May 18 06:57:25.954: ISDN Se0/0/1:15 Q931: TX -> STATUS pd = 8 callref = 0x8E82 Cause i = 0x80E20F - Message not compatible with call state or not implemented Call State i = 0x13 May 18 06:57:25.962: ISDN Se0/0/1:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x0E82 Cause i = 0x82AF - Resource unavailable, unspecified May 18 06:57:25.966: ISDN Se0/0/1:15 Q931: RX <- RESTART pd = 8 callref = 0x0000 Channel ID i = 0xA98385 Exclusive, Channel 5 Restart Indicator i = 0x80 May 18 06:57:25.966: ISDN Se0/0/1:15 Q931: TX -> RESTART_ACK pd = 8 callref = 0x8000 Channel ID i = 0xA98385 Exclusive, Channel 5 Restart Indicator i = 0x80 May 18 06:57:25.978: ISDN Se0/0/1:15 Q931: RX <- SETUP pd = 8 callref = 0x0E83 Sending Complete Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98395 Exclusive, Channel 21 Calling Party Number i = 0x2183, '730730199' Plan:ISDN, Type:National Called Party Number i = 0xA1, '730732888' Plan:ISDN, Type:National May 18 06:57:25.978: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0E82 Cause i = 0x80E5 - Message not compatible with call state May 18 06:57:25.982: ISDN Se0/0/1:15 **ERROR**: L3_BadPeerMsg: event 0x5A cr 0x8E82 callid 0x0 OULKLAVG2# May 18 06:57:25.982: ISDN Se0/0/1:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8E83 Channel ID i = 0xA98395 Exclusive, Channel 21 OULKLAVG2# May 18 06:57:29.054: ISDN Se0/0/1:15 Q931: TX -> CONNECT pd = 8 callref = 0x8E83 May 18 06:57:29.066: ISDN Se0/0/1:15 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x0E83 May 18 06:57:29.066: %ISDN-6-CONNECT: Interface Serial0/0/1:20 is now connected to 730730199 N/A May 18 06:57:29.206: ISDN Se0/0/1:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x0E83 Cause i = 0x82AF - Resource unavailable, unspecified May 18 06:57:29.206: %ISDN-6-CONNECT: Interface Serial0/0/1:20 is now connected to 730730199 N/A OULKLAVG2# May 18 06:57:29.206: ISDN Se0/0/1:15 Q931: TX -> RELEASE pd = 8 callref = 0x8E83 May 18 06:57:29.210: ISDN Se0/0/1:15 Q931: RX <- RESTART pd = 8 callref = 0x0000 Channel ID i = 0xA98395 Exclusive, Channel 21 Restart Indicator i = 0x80 May 18 06:57:29.210: ISDN Se0/0/1:15 Q931: TX -> RESTART_ACK pd = 8 callref = 0x8000 Channel ID i = 0xA98395 Exclusive, Channel 21 Restart Indicator i = 0x80 May 18 06:57:29.218: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0E83 Cause i = 0x80D1 - Invalid call reference value May 18 06:57:29.218: ISDN Se0/0/1:15 **ERROR**: L3_BadPeerMsg: event 0x5A cr 0x8E83 callid 0x0 Cause i = 0x82AF - Resource unavailable, unspecified May 18 06:57:25.930: %ISDN-6-CONNECT: Interface Serial0/0/1:4 is now connected to 730730199 N/A May 18 06:57:25.930: ISDN Se0/0/1:15 Q931: TX -> RELEASE pd = 8 callref = 0x8E82 May 18 06:57:25.942: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0E82 Cause i = 0x80D1 - Invalid call reference value May 18 06:57:25.954: ISDN Se0/0/1:15 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x0E82 May 18 06:57:25.954: ISDN Se0/0/1:15 **ERROR**: Ux_BadMsg: Invalid Message for call state 19, call id 0x3611, call ref 0x8E82, event 0xF May 18 06:57:25.954: ISDN Se0/0/1:15 Q931: TX -> STATUS pd = 8 callref = 0x8E82 Cause i = 0x80E20F - Message not compatible with call state or not implemented Call State i = 0x13 May 18 06:57:25.962: ISDN Se0/0/1:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x0E82 Cause i = 0x82AF - Resource unavailable, unspecified May 18 06:57:25.966: ISDN Se0/0/1:15 Q931: RX <- RESTART pd = 8 callref = 0x0000 Channel ID i = 0xA98385 Exclusive, Channel 5 Restart Indicator i = 0x80 May 18 06:57:25.966: ISDN Se0/0/1:15 Q931: TX -> RESTART_ACK pd = 8 callref = 0x8000 Channel ID i = 0xA98385 Exclusive, Channel 5 Restart Indicator i = 0x80 May 18 06:57:25.978: ISDN Se0/0/1:15 Q931: RX <- SETUP pd = 8 callref = 0x0E83 Sending Complete Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98395 Exclusive, Channel 21 Calling Party Number i = 0x2183, '730730199' Plan:ISDN, Type:National Called Party Number i = 0xA1, '730732888' Plan:ISDN, Type:National May 18 06:57:25.978: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0E82 Cause i = 0x80E5 - Message not compatible with call state May 18 06:57:25.982: ISDN Se0/0/1:15 **ERROR**: L3_BadPeerMsg: event 0x5A cr 0x8E82 callid 0x0 OULKLAVG2# May 18 06:57:25.982: ISDN Se0/0/1:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8E83 Channel ID i = 0xA98395 Exclusive, Channel 21 OULKLAVG2# May 18 06:57:29.054: ISDN Se0/0/1:15 Q931: TX -> CONNECT pd = 8 callref = 0x8E83 May 18 06:57:29.066: ISDN Se0/0/1:15 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x0E83 May 18 06:57:29.066: %ISDN-6-CONNECT: Interface Serial0/0/1:20 is now connected to 730730199 N/A May 18 06:57:29.206: ISDN Se0/0/1:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x0E83 Cause i = 0x82AF - Resource unavailable, unspecified May 18 06:57:29.206: %ISDN-6-CONNECT: Interface Serial0/0/1:20 is now connected to 730730199 N/A OULKLAVG2# May 18 06:57:29.206: ISDN Se0/0/1:15 Q931: TX -> RELEASE pd = 8 callref = 0x8E83 May 18 06:57:29.210: ISDN Se0/0/1:15 Q931: RX <- RESTART pd = 8 callref = 0x0000 Channel ID i = 0xA98395 Exclusive, Channel 21 Restart Indicator i = 0x80 May 18 06:57:29.210: ISDN Se0/0/1:15 Q931: TX -> RESTART_ACK pd = 8 callref = 0x8000 Channel ID i = 0xA98395 Exclusive, Channel 21 Restart Indicator i = 0x80 May 18 06:57:29.218: ISDN Se0/0/1:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x0E83 Cause i = 0x80D1 - Invalid call reference value May 18 06:57:29.218: ISDN Se0/0/1:15 **ERROR**: L3_BadPeerMsg: event 0x5A cr 0x8E83 callid 0x0
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